Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 5a0dc13a6d7025e95723482d566e5477b35b6c83..9966c4ab6312c708a075c9a55e7762e91b41d7e4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -29,6 +29,8 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+
 namespace webrtc {
 
 namespace {
@@ -904,6 +906,22 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
   }
   packet->SetExtension<AbsoluteSendTime>(now_ms);
 
+  if (video_) {
+    BWE_TEST_LOGGING_PLOT(1, "SentBitrate[Kbps]", \
+      now_ms, ActualSendBitrateKbit());
+    BWE_TEST_LOGGING_PLOT(1, "FecBitrate[Kbps]", \
+      now_ms, FecOverheadRate()/1000);
+    BWE_TEST_LOGGING_PLOT(1, "NackBitrate[Kbps]", \
+      now_ms, NackOverheadRate()/1000);
+    BWE_TEST_LOGGING_PLOT(1, "VideoBitrate[bps]", \
+      now_ms, VideoBitrateSent()/1000);
+  }
+
+  if (!video_) {
+    BWE_TEST_LOGGING_PLOT(1, "AudioSentBitrate[Kbps]", \
+      now_ms, ActualSendBitrateKbit());
+  }
+

[stefan-webrtc, 2016/09/01 14:07:36]

Note that these are measured per ssrc, so I think you have to add ssrc to the
logs. Or if you are only interested in per transport you could choose to do
something different. Maybe we're not even interested in knowing if it's audio,
video or retransmits etc?

[Gaetano Carlucci, 2016/09/01 16:06:26]

yes, adding ssrc should be enough 

   if (paced_sender_) {
     uint16_t seq_no = packet->SequenceNumber();
     uint32_t ssrc = packet->Ssrc();